[Asterisk-Dev] Digium's G.729A codec problem

[Daniel Pocock]

The Digium implementation is closed source :( unlike the rest of 
Asterisk, so you probably won't be able to troubleshoot this yourself.

Try the open source implementation and let me know if you have the same 
problem:

    http://www.readytechnology.co.uk/open/g729

Make sure you are watching the Asterisk console when you try making 
calls, my code spits out error messages if packets are the wrong size, 
etc.  This will give you some clues.

Regards,

Daniel

Jacky wrote:

>Hi, all,
>
>I have buy 5 Digium's G.729A codec(it just support G.729A license)
>When I  calll with 2 SIP UA that support G.729A and G.729B, its rtp frame 
>have some problem when softswitch with Asterisk.
>
>The voice frame have been drop, so sometime I can't hear voice.
>
>If I want to fix the problem when softswitch G.729A and G.729B codec.
>What source code I must to modify ?
>Or some people have finished the issue, Could you show me how to do?
>
> 
>  
>